Asterisk for a Telephone Answering Service

Hello All,
I have followed Asterisk throughout the years and there are a great number of features that we could use in an Answering Service environment right out of the box (call queues, call recording, routing, caller-id, etc.). However an Answering Service differs from a “call center” in that we have customized instructions that change for clients on a regular basis. We maintain on-call schedules and appropriate message delivery (email, fax SMS) based on time and priority.

We have enough experience with web development to build the client form integration (database lookups based on account information), however its the bridge between the Asterisk call queue and the individual operator stations that I am drawing a blank.

Basically we need a round robin call queue and the ability for operators to stack up to 6 calls at their station for parallel processing. Based on the number dialed we would need this to do a database lookup of the customers instructions for that account. Is there a simple way to hold a call in the queue and have the operator “flip” it to their station with the DID (Direct Inward Dial) number information. This number is used as the account number for database lookups.

This is alot of information to post here, so I can get into more specifics. There are some posts here about using Asterisk as a TAS, however there doesn’t seem to be an specific direction to accomplish this piece of the puzzle.

Any guidance would be greatly appreciate, either Open Source or commercial.



You can grab calls out of a queue and send them to a station using the Asterisk Manager Interface. I’m not sure I understand what you mean by “stack up 6 calls”, though. Do you mean have a main queue and then a 6-deep subqueue for each station? You can move calls around those queues with AMI as well.

I think that you are speaking with a FreePBX installation in mind. Asterisk is just a sip server with no configuration, so DIDs etc are not something that Asterisk understands. After getting this out of the way let’s see what FreePBX does.
When a call comes sets the DID variable, this variable can be used throughout the call. So if you do the same in your Asterisk installation, set a variable with __ at the begining you will make the variable accesible until the end of the call.
This means that when you receive a call from the queue you will have the information that you want already as a channel variable. I think you should give more details on your project because from the first post the only think that we can do is imagine what do you want to do. Asterisk is a white canvas so without any details you will not get very far.

Oh come on, these boards give me headache I thought I was responding to a post not necrobumping an old thread! Admins please delete if you think is bad.